[Freeswitch-users] Silence suppression again
Alex Lake
alex at digitalmail.com
Fri Mar 1 12:26:38 MSK 2013
Done them!
> Alex,
>
> I suspect these pages would be a good place to start:
> http://wiki.freeswitch.org/wiki/VAD_and_CNG
> http://wiki.freeswitch.org/wiki/Variable_suppress_cng
>
> -MC
>
> On Thu, Feb 28, 2013 at 12:57 AM, Alex Lake <alex at digitalmail.com
> <mailto:alex at digitalmail.com>> wrote:
>
> Ah! I think that has fixed it. Many thanks. I'll see if I can
> update a bit of wiki somewhere...
>> suppress_cng=true variable or supress-cng profile param set to true.
>>
>>
>> On Wed, Feb 27, 2013 at 11:44 AM, Alex Lake <alex at digitalmail.com
>> <mailto:alex at digitalmail.com>> wrote:
>>
>> A quick update on this - I have verified that suppresion of
>> silence suppression(!) on Grandstreams doesn't work when it
>> is the B-Party of a call. I'm not sure if silence-suppression
>> is part of the SIP handshake, but I know that other VOIP
>> technologies seem to be able to prevent it.
>> I'm calling in a handset specialist to advise..
>>
>> One of the things that intrigues me is this extract from the
>> log (particularly the reference to silenceSupp):
>>
>> 2013-02-27 17:28:05.151624 [DEBUG] switch_core_codec.c:244
>> sofia/internal/0253302 at 004-0253.sb12.dmclub.org
>> <mailto:sofia/internal/0253302 at 004-0253.sb12.dmclub.org>
>> Restore previous codec PCMA:8.
>> 2013-02-27 17:28:05.151624 [DEBUG] mod_sofia.c:754 Local SDP
>> sofia/internal/0253302 at 004-0253.sb12.dmclub.org
>> <mailto:sofia/internal/0253302 at 004-0253.sb12.dmclub.org>:
>> v=0
>> o=FreeSWITCH 1361975871 1361975873 IN IP4 176.58.88.201
>> s=FreeSWITCH
>> c=IN IP4 176.58.88.201
>> t=0 0
>> m=audio 10212 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:30
>> a=sendrecv
>>
>> What does all this mean?
>>
>> From the "A" leg of the call (which is placed by a LinkSys
>> SPA922) I have:
>>
>> <sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201361974765%201361974767%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011468%20RTP/AVP%208%20101%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
>>
>> From the troublesome "B" leg of the call (placed by
>> FreeSwitch to my GXP2000) I have:
>>
>> sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201361974745%201361974746%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011488%20RTP/AVP%208%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0Am%3Daudio%2011488%20RTP/AVP%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</sip_local_sdp_str>
>>
>> I was wondering if there's a way I can modify this to include
>> a="silenceSupp:off - - - -" and whether that would make the
>> slightest bit of difference to the handset's behaviour!
>>
>>
>>
>> Alex
>>> Thanks.
>>> I've been looking at the channel variables of the call. And
>>> this is part of it:
>>>
>>> <rtp_audio_in_raw_bytes>1089091</rtp_audio_in_raw_bytes><rtp_audio_in_media_bytes>957212</rtp_audio_in_media_bytes><rtp_audio_in_packet_count>10344</rtp_audio_in_packet_count><rtp_audio_in_media_packet_count>5581</rtp_audio_in_media_packet_count><rtp_audio_in_skip_packet_count>5480</rtp_audio_in_skip_packet_count><rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count><rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count><rtp_audio_in_cng_packet_count>4323</rtp_audio_in_cng_packet_count><rtp_audio_in_flush_packet_count>440</rtp_audio_in_flush_packet_count><rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size><rtp_audio_out_raw_bytes>1829908</rtp_audio_out_raw_bytes><rtp_audio_out_media_bytes>1829908</rtp_audio_out_media_bytes><rtp_audio_out_packet_count>10639</rtp_audio_out_packet_count><rtp_audio_out
>>> _media_packet_count>10639</rtp_audio_out_media_packet_count><rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count><rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count><rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count><rtp_audio_rtcp_packet_count>0</rtp_audio_rtcp_packet_count><rtp_audio_rtcp_octet_count>0</rtp_audio_rtcp_octet_count>
>>>
>>> The thing that slightly concerns me here is
>>> <rtp_audio_in_cng_packet_count>4323</rtp_audio_in_cng_packet_count>
>>>
>>> That sounds as though the inbound audio path contains some
>>> comfort noise, suggesting that the handset the other end has
>>> some kind of silence suppression enabled.
>>>
>>> I'm afraid it's a dreaded Grandstream GXP2000 - which I know
>>> are not the handset of the cognoscenti! - but the customer
>>> tells me that he has silence suppression disabled. Do we
>>> think this is correct?!
>>>
>>> Rgds,
>>> Alex
>>>> It does not send any by default. Also, yes it only
>>>> generates silence when the call is not getting any RTP.
>>>>
>>>>
>>>>
>>>> On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake
>>>> <alex at digitalmail.com <mailto:alex at digitalmail.com>> wrote:
>>>>
>>>> Does Freeswitch enable any kind of silence suppression
>>>> by default? If I
>>>> have bridge_generate_comfort_noise=true, does that only
>>>> have an effect
>>>> when the audio-generating end decides to stop sending RTP?
>>>>
>>>> Any other tips for diagnosing silence (it's kind of
>>>> like a slow noise
>>>> gate effect)?
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
>>>> Official FreeSWITCH Sites
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>>>>
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>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
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>>>> <mailto:sip%3A888 at conference.freeswitch.org>
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>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900 <tel:%2B19193869900>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
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>>>>
>>>> No virus found in this message.
>>>> Checked by AVG - www.avg.com <http://www.avg.com>
>>>> Version: 2012.0.2238 / Virus Database: 2641/5634 - Release
>>>> Date: 02/26/13
>>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
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>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
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>>>
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>>> Checked by AVG - www.avg.com <http://www.avg.com>
>>> Version: 2012.0.2238 / Virus Database: 2641/5635 - Release
>>> Date: 02/26/13
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900 <tel:%2B19193869900>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
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>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com <http://www.avg.com>
>> Version: 2012.0.2238 / Virus Database: 2641/5636 - Release Date:
>> 02/27/13
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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> http://www.freeswitch.org
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2238 / Virus Database: 2641/5638 - Release Date: 02/28/13
>
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